tg2sip/webrtc_dsp/modules/audio_processing/aec3/moving_average.cc

61 lines
1.7 KiB
C++

/*
* Copyright (c) 2018 The WebRTC project authors. All Rights Reserved.
*
* Use of this source code is governed by a BSD-style license
* that can be found in the LICENSE file in the root of the source
* tree. An additional intellectual property rights grant can be found
* in the file PATENTS. All contributing project authors may
* be found in the AUTHORS file in the root of the source tree.
*/
#include "modules/audio_processing/aec3/moving_average.h"
#include <algorithm>
#include <functional>
#include "rtc_base/checks.h"
namespace webrtc {
namespace aec3 {
MovingAverage::MovingAverage(size_t num_elem, size_t mem_len)
: num_elem_(num_elem),
mem_len_(mem_len - 1),
scaling_(1.0f / static_cast<float>(mem_len)),
memory_(num_elem * mem_len_, 0.f),
mem_index_(0) {
RTC_DCHECK(num_elem_ > 0);
RTC_DCHECK(mem_len > 0);
}
MovingAverage::~MovingAverage() = default;
void MovingAverage::Average(rtc::ArrayView<const float> input,
rtc::ArrayView<float> output) {
RTC_DCHECK(input.size() == num_elem_);
RTC_DCHECK(output.size() == num_elem_);
// Sum all contributions.
std::copy(input.begin(), input.end(), output.begin());
for (auto i = memory_.begin(); i < memory_.end(); i += num_elem_) {
std::transform(i, i + num_elem_, output.begin(), output.begin(),
std::plus<float>());
}
// Divide by mem_len_.
for (float& o : output) {
o *= scaling_;
}
// Update memory.
if (mem_len_ > 0) {
std::copy(input.begin(), input.end(),
memory_.begin() + mem_index_ * num_elem_);
mem_index_ = (mem_index_ + 1) % mem_len_;
}
}
} // namespace aec3
} // namespace webrtc